The use of radio frequency media for telephony has become widely available in recent years. A digital cordless telephone (Second Generation Cordless Telephone: CT-2) system has recently been introduced into the market and the present-day CT-2 system is regarded as an access point in a PSTN (public switched telephone network).
Such a CT-2 system comprises a base station connected to the PSTN for communicating with other PSTN subscriber, wherein, by means of the base station, only outward calls are possible for a CT-2 terminal. And the CT-2 system further comprises a subscriber managing unit connected to a PSDN (public switched data network) for communicating with the base station to authenticate the CT-2 terminal.
An example of the conventional CT-2 system 100 is described in FIGS. 1 and 2. In FIG. 1, there is provided a block diagram of a conventional CT-2 system 100. In FIG. 2, there is provided a block diagram of the base station 10 of the CT-2 system 100 shown in FIG. 1.
Referring to FIG. 1, the CT-2 system 100 comprises a plurality of CT-2 terminals (handsets: HS's) 1 to 4, a multiplicity of base stations (BS's) 10 to 14, a base station connecting (BSC) unit 20, and a subscriber managing (SM) unit 30.
Each of the HS's 1 to 4 is connected to one of BS's 10 to 14 by radio channels, using a known digital signaling protocol, e.g., the CT-2 CAI (Common Air Interface) signaling protocol. And each of the BS's 10 to 1 is coupled with the PSTN 40 by a predetermined number, e.g., four, of two-wired cables, wherein each of the four cables has the capability of an analog subscriber connection. And also, each of the BS's 10 to 14 is coupled with the BSC unit 20 by a two-wired cable for predetermined data transmission.
Hereafter, for the sake of illustration, it is assumed that a call is originated by the HS 1 and that the HS 1 is within the serving area of the BS 10. When the call is originated by the HS 1, a call origination message including a PID (Portable Identity Code) of the HS 1 is transmitted to the BS 10. Then, a radio channel is assigned by the BS 10 and the PID is transferred from the BS 10 to the BSC unit 20. At the BSC unit 20, the PID is formed into a predetermined protocol format, e.g., X.25 protocol packet, to be transferred to the SM unit 30 for terminal authentication through the PSDN 50.
If the HS 1 is authenticated as a registered one by the SM unit 30, a channel connected to the PSTN 40 is assigned to the HS 1 by the BS 10. And then, the destination subscriber number is dialed by the HS 1 to be transferred to the PSTN 40.
Since the rest of the HS's 2 to 4 and the rest of the BS's 12 to 14 are substantially identical to the HS 1 and the BS 10, respectively, further explanation on their operation is omitted here for the sake of simplicity.
Referring now to FIG. 2, the BS 10 includes a radio frequency interfacing (RFI) unit 22, four ADPCM/PCM (Adaptive Differential Pulse-Code Modulation/Pulse-Code Modulation) converting (APC) units 32 to 38, four codec/filters 42 to 48, four analog subscriber interfacing (ASI) units 52 to 58, a controller 62, a modem interfacing (MI) unit 72.
In the HS 1, speech is modulated to a 32 kbps ADPCM signal and a predetermined CT-2 CAI message required while the call is generated. The 32 kbps up-stream ADPCM signal is transmitted via a speech channel, and the digital signaling message via a control channel.
The RFI unit 22 may be a typical radio frequency transceiver. Through the RFI unit 22, the 32 kbps up-stream ADPCM signal is applied to one of the APC units, e.g., the APC unit 32; and the CT-2 CAI message is applied to the controller 62.
In the APC unit 32, the 32 kbps up-stream ADPCM signal is converted to a 64 kbps PCM signal. The converted 64 kbps PCM signal is applied to the corresponding codec/filer 42. The 64 kbps PCM signal is decoded and filtered to generate an analog speech signal at the codec/filter 42.
The CT-2 CAI message is analyzed by the controller 62 which controls the ASI unit 52 for transferring the analog speech signal and the predetermined subscriber line signaling, e.g., the hook-on/off, dialing and the like, to the PSTN 40 and also monitors the ASI unit 52 for the channel status, e.g., channel seizure status and the like.
On the other hand, the controller 62 is coupled with the MI unit 72 to communicate with the SM unit 30 for terminal authentication. The MI unit 72 may be a typical modem circuit. Data for the terminal authentication, including the PID of the HS 1, is transferred synchronously at the transfer speed, e.g., of 1.2 kbps from the MI unit 72 to the BSC unit 20.
The detailed description of the rest of the APC's 34 to 38, the codec/filters 44 to 48, the ASI's 54 to 58 is omitted here for the same reason as the HS's 2 to 4. And also, since the description for the down-stream signal is well known to a skilled person in the art, it is omitted here for the sake of brevity.
However, in case of the prior art, the cost to construct the entire CT-2 system 100 is very high since each of the ASI's 52 to 58 is connected to the PSTN by its own two-wired cables. In other words, non-switched/non-concentrated channels tend to entail a low efficiency in using the subscriber circuit (not shown) within the PSTN 40.
Further, the call authentication is delayed and the connectivity of the call is degraded because the data is transferred synchronously at a low transfer speed between each of the BS's 10 to 14 and the BSC unit 20 via the modem.
Furthermore, since the speech is converted to a digital signal in each of the HS's 1 to 4 and converted again to an analog signal at the BS 10, it is complicated to configure, operate and maintain the CT-2 system 100. And also, the quality of the call may be degraded due to the multi conversion processes.
Consequently, it is desirable to provide a cordless telephone system capable of switching and concentrating channels between each of the BS's 10 to 14 and the PSTN 40 and improving the connectivity and quality of a call.